upload p3

checkpoints/FastDiff/config.yaml

@@ -0,0 +1,149 @@
+N: ''
+T: 1000
+accumulate_grad_batches: 1
+amp: false
+audio_channels: 1
+audio_num_mel_bins: 80
+audio_sample_rate: 22050
+aux_context_window: 0
+beta_0: 1.0e-06
+beta_T: 0.01
+binarization_args:
+  reset_phone_dict: true
+  reset_word_dict: true
+  shuffle: false
+  trim_eos_bos: false
+  with_align: false
+  with_f0: false
+  with_f0cwt: false
+  with_linear: false
+  with_spk_embed: false
+  with_spk_id: true
+  with_txt: false
+  with_wav: true
+  with_word: false
+binarizer_cls: data_gen.tts.vocoder_binarizer.VocoderBinarizer
+binary_data_dir: data/binary/LJSpeech
+check_val_every_n_epoch: 10
+clip_grad_norm: 1
+clip_grad_value: 0
+cond_channels: 80
+debug: false
+dec_ffn_kernel_size: 9
+dec_layers: 4
+dict_dir: ''
+diffusion_step_embed_dim_in: 128
+diffusion_step_embed_dim_mid: 512
+diffusion_step_embed_dim_out: 512
+disc_start_steps: 40000
+discriminator_grad_norm: 1
+dropout: 0.0
+ds_workers: 1
+enc_ffn_kernel_size: 9
+enc_layers: 4
+endless_ds: true
+eval_max_batches: -1
+ffn_act: gelu
+ffn_padding: SAME
+fft_size: 1024
+fmax: 7600
+fmin: 80
+frames_multiple: 1
+gen_dir_name: ''
+generator_grad_norm: 10
+griffin_lim_iters: 60
+hidden_size: 256
+hop_size: 256
+infer: false
+inner_channels: 32
+kpnet_conv_size: 3
+kpnet_hidden_channels: 64
+load_ckpt: ''
+loud_norm: false
+lr: 2e-4
+lvc_kernel_size: 3
+lvc_layers_each_block: 4
+max_epochs: 1000
+max_frames: 1548
+max_input_tokens: 1550
+max_samples: 25600
+max_sentences: 20
+max_tokens: 30000
+max_updates: 1000000
+max_valid_sentences: 1
+max_valid_tokens: 60000
+mel_loss: l1
+mel_vmax: 1.5
+mel_vmin: -6
+mfa_version: 2
+min_frames: 0
+min_level_db: -100
+noise_schedule: ''
+num_ckpt_keep: 3
+num_heads: 2
+num_mels: 80
+num_sanity_val_steps: -1
+num_spk: 400
+num_test_samples: 0
+num_valid_plots: 10
+optimizer_adam_beta1: 0.9
+optimizer_adam_beta2: 0.98
+out_wav_norm: false
+pitch_extractor: parselmouth
+pre_align_args:
+  allow_no_txt: false
+  denoise: false
+  nsample_per_mfa_group: 1000
+  sox_resample: false
+  sox_to_wav: false
+  trim_sil: false
+  txt_processor: en
+  use_tone: true
+pre_align_cls: egs.datasets.audio.pre_align.PreAlign
+print_nan_grads: false
+processed_data_dir: data/processed/LJSpeech
+profile_infer: false
+raw_data_dir: data/raw/LJSpeech-1.1
+ref_level_db: 20
+rename_tmux: true
+resume_from_checkpoint: 0
+save_best: true
+save_codes: []
+save_f0: false
+save_gt: true
+scheduler: rsqrt
+seed: 1234
+sort_by_len: true
+task_cls: modules.FastDiff.task.FastDiff.FastDiffTask
+tb_log_interval: 100
+test_ids: []
+test_input_dir: ''
+test_mel_dir: ''
+test_num: 100
+test_set_name: test
+train_set_name: train
+train_sets: ''
+upsample_ratios:
+- 8
+- 8
+- 4
+use_pitch_embed: false
+use_spk_embed: false
+use_spk_id: false
+use_split_spk_id: false
+use_wav: true
+use_weight_norm: true
+use_word_input: false
+val_check_interval: 2000
+valid_infer_interval: 10000
+valid_monitor_key: val_loss
+valid_monitor_mode: min
+valid_set_name: valid
+vocoder_denoise_c: 0.0
+warmup_updates: 8000
+weight_decay: 0
+win_length: null
+win_size: 1024
+window: hann
+word_size: 30000
+work_dir: checkpoints/FastDiff

checkpoints/FastDiff/model_ckpt_steps_500000.ckpt

@@ -0,0 +1,3 @@
+version https://git-lfs.github.com/spec/v1
+oid sha256:ee7b6022e525c71a6025b41eeeafff9d6186b52cba76b580d6986bc8674902f3
+size 183951271

checkpoints/ProDiff/config.yaml

@@ -0,0 +1,205 @@
+accumulate_grad_batches: 1
+amp: false
+audio_num_mel_bins: 80
+audio_sample_rate: 22050
+base_config:
+- ./base.yaml
+binarization_args:
+  reset_phone_dict: true
+  reset_word_dict: true
+  shuffle: false
+  trim_eos_bos: false
+  trim_sil: false
+  with_align: true
+  with_f0: true
+  with_f0cwt: false
+  with_linear: false
+  with_spk_embed: false
+  with_spk_id: true
+  with_txt: true
+  with_wav: false
+  with_word: true
+binarizer_cls: data_gen.tts.base_binarizer.BaseBinarizer
+binary_data_dir: data/binary/LJSpeech
+check_val_every_n_epoch: 10
+clip_grad_norm: 1
+clip_grad_value: 0
+conv_use_pos: false
+cwt_add_f0_loss: false
+cwt_hidden_size: 128
+cwt_layers: 2
+cwt_loss: l1
+cwt_std_scale: 0.8
+debug: false
+dec_dilations:
+- 1
+- 1
+- 1
+- 1
+dec_ffn_kernel_size: 9
+dec_inp_add_noise: false
+dec_kernel_size: 5
+dec_layers: 4
+dec_num_heads: 2
+decoder_rnn_dim: 0
+decoder_type: fft
+dict_dir: ''
+diff_decoder_type: wavenet
+diff_loss_type: l1
+dilation_cycle_length: 1
+dropout: 0.1
+ds_workers: 2
+dur_enc_hidden_stride_kernel:
+- 0,2,3
+- 0,2,3
+- 0,1,3
+dur_loss: mse
+dur_predictor_kernel: 3
+dur_predictor_layers: 2
+enc_dec_norm: ln
+enc_dilations:
+- 1
+- 1
+- 1
+- 1
+enc_ffn_kernel_size: 9
+enc_kernel_size: 5
+enc_layers: 4
+encoder_K: 8
+encoder_type: fft
+endless_ds: true
+ffn_act: gelu
+ffn_hidden_size: 1024
+ffn_padding: SAME
+fft_size: 1024
+fmax: 7600
+fmin: 80
+frames_multiple: 1
+gen_dir_name: ''
+gen_tgt_spk_id: -1
+griffin_lim_iters: 60
+hidden_size: 256
+hop_size: 256
+infer: false
+keep_bins: 80
+lambda_commit: 0.25
+lambda_energy: 0.1
+lambda_f0: 1.0
+lambda_ph_dur: 0.1
+lambda_sent_dur: 1.0
+lambda_uv: 1.0
+lambda_word_dur: 1.0
+layers_in_block: 2
+load_ckpt: ''
+loud_norm: false
+lr: 1.0
+max_beta: 0.06
+max_epochs: 1000
+max_frames: 1548
+max_input_tokens: 1550
+max_sentences: 48
+max_tokens: 32000
+max_updates: 200000
+max_valid_sentences: 1
+max_valid_tokens: 60000
+mel_loss: ssim:0.5|l1:0.5
+mel_vmax: 1.5
+mel_vmin: -6
+min_frames: 0
+min_level_db: -100
+num_ckpt_keep: 3
+num_heads: 2
+num_sanity_val_steps: -1
+num_spk: 400
+num_test_samples: 0
+num_valid_plots: 10
+optimizer_adam_beta1: 0.9
+optimizer_adam_beta2: 0.98
+out_wav_norm: false
+pitch_ar: false
+pitch_embed_type: 0
+pitch_enc_hidden_stride_kernel:
+- 0,2,5
+- 0,2,5
+- 0,2,5
+pitch_extractor: parselmouth
+pitch_loss: l1
+pitch_norm: standard
+pitch_ssim_win: 11
+pitch_type: frame
+pre_align_args:
+  allow_no_txt: false
+  denoise: false
+  sox_resample: false
+  sox_to_wav: false
+  trim_sil: false
+  txt_processor: en
+  use_tone: true
+pre_align_cls: ''
+predictor_dropout: 0.5
+predictor_grad: 0.1
+predictor_hidden: -1
+predictor_kernel: 5
+predictor_layers: 2
+pretrain_fs_ckpt: ''
+print_nan_grads: false
+processed_data_dir: data/processed/LJSpeech
+profile_infer: false
+raw_data_dir: data/raw/LJSpeech
+ref_hidden_stride_kernel:
+- 0,3,5
+- 0,3,5
+- 0,2,5
+- 0,2,5
+- 0,2,5
+ref_level_db: 20
+ref_norm_layer: bn
+rename_tmux: true
+residual_channels: 256
+residual_layers: 20
+resume_from_checkpoint: 0
+save_best: true
+save_codes: []
+save_f0: false
+save_gt: true
+schedule_type: vpsde
+scheduler: rsqrt
+seed: 1234
+sil_add_noise: false
+sort_by_len: true
+spec_max: []
+spec_min: []
+task_cls: modules.ProDiff.task.ProDiff_task.ProDiff_Task
+tb_log_interval: 100
+teacher_ckpt: checkpoints/ProDiff_Teacher/model_ckpt_steps_188000.ckpt
+test_ids: []
+test_input_dir: ''
+test_num: 100
+test_set_name: test
+timesteps: 4
+train_set_name: train
+train_sets: ''
+use_cond_disc: true
+use_energy_embed: true
+use_gt_dur: true
+use_gt_f0: true
+use_pitch_embed: true
+use_pos_embed: true
+use_ref_enc: false
+use_spk_embed: false
+use_spk_id: true
+use_split_spk_id: false
+use_uv: true
+use_var_enc: false
+val_check_interval: 2000
+valid_infer_interval: 10000
+valid_monitor_key: val_loss
+valid_monitor_mode: min
+valid_set_name: valid
+var_enc_vq_codes: 64
+vocoder_denoise_c: 0.0
+warmup_updates: 2000
+weight_decay: 0
+win_size: 1024
+word_size: 30000
+work_dir: checkpoints/ProDiff

checkpoints/ProDiff/model_ckpt_steps_200000.ckpt

@@ -0,0 +1,3 @@
+version https://git-lfs.github.com/spec/v1
+oid sha256:8cc8aad355c297b010e2c362341f736b3477744af76e02f6c9965409a7e9113a
+size 349055740

checkpoints/ProDiff_Teacher/config.yaml

@@ -0,0 +1,205 @@
+accumulate_grad_batches: 1
+amp: false
+audio_num_mel_bins: 80
+audio_sample_rate: 22050
+base_config:
+- ./base.yaml
+binarization_args:
+  reset_phone_dict: true
+  reset_word_dict: true
+  shuffle: false
+  trim_eos_bos: false
+  trim_sil: false
+  with_align: true
+  with_f0: true
+  with_f0cwt: false
+  with_linear: false
+  with_spk_embed: false
+  with_spk_id: true
+  with_txt: true
+  with_wav: false
+  with_word: true
+binarizer_cls: data_gen.tts.base_binarizer.BaseBinarizer
+binary_data_dir: data/binary/LJSpeech
+check_val_every_n_epoch: 10
+clip_grad_norm: 1
+clip_grad_value: 0
+conv_use_pos: false
+cwt_add_f0_loss: false
+cwt_hidden_size: 128
+cwt_layers: 2
+cwt_loss: l1
+cwt_std_scale: 0.8
+debug: false
+dec_dilations:
+- 1
+- 1
+- 1
+- 1
+dec_ffn_kernel_size: 9
+dec_inp_add_noise: false
+dec_kernel_size: 5
+dec_layers: 4
+dec_num_heads: 2
+decoder_rnn_dim: 0
+decoder_type: fft
+dict_dir: ''
+diff_decoder_type: wavenet
+diff_loss_type: l1
+dilation_cycle_length: 1
+dropout: 0.1
+ds_workers: 2
+dur_enc_hidden_stride_kernel:
+- 0,2,3
+- 0,2,3
+- 0,1,3
+dur_loss: mse
+dur_predictor_kernel: 3
+dur_predictor_layers: 2
+enc_dec_norm: ln
+enc_dilations:
+- 1
+- 1
+- 1
+- 1
+enc_ffn_kernel_size: 9
+enc_kernel_size: 5
+enc_layers: 4
+encoder_K: 8
+encoder_type: fft
+endless_ds: true
+ffn_act: gelu
+ffn_hidden_size: 1024
+ffn_padding: SAME
+fft_size: 1024
+fmax: 7600
+fmin: 80
+frames_multiple: 1
+gen_dir_name: ''
+gen_tgt_spk_id: -1
+griffin_lim_iters: 60
+hidden_size: 256
+hop_size: 256
+infer: false
+keep_bins: 80
+lambda_commit: 0.25
+lambda_energy: 0.1
+lambda_f0: 1.0
+lambda_ph_dur: 0.1
+lambda_sent_dur: 1.0
+lambda_uv: 1.0
+lambda_word_dur: 1.0
+layers_in_block: 2
+load_ckpt: ''
+loud_norm: false
+lr: 1.0
+max_beta: 0.06
+max_epochs: 1000
+max_frames: 1548
+max_input_tokens: 1550
+max_sentences: 48
+max_tokens: 32000
+max_updates: 200000
+max_valid_sentences: 1
+max_valid_tokens: 60000
+mel_loss: ssim:0.5|l1:0.5
+mel_vmax: 1.5
+mel_vmin: -6
+min_frames: 0
+min_level_db: -100
+num_ckpt_keep: 3
+num_heads: 2
+num_sanity_val_steps: -1
+num_spk: 400
+num_test_samples: 20
+num_valid_plots: 10
+optimizer_adam_beta1: 0.9
+optimizer_adam_beta2: 0.98
+out_wav_norm: false
+pitch_ar: false
+pitch_embed_type: 0
+pitch_enc_hidden_stride_kernel:
+- 0,2,5
+- 0,2,5
+- 0,2,5
+pitch_extractor: parselmouth
+pitch_loss: l1
+pitch_norm: standard
+pitch_ssim_win: 11
+pitch_type: frame
+pre_align_args:
+  allow_no_txt: false
+  denoise: false
+  sox_resample: false
+  sox_to_wav: false
+  trim_sil: false
+  txt_processor: en
+  use_tone: true
+pre_align_cls: egs.datasets.audio.lj.pre_align.LJPreAlign
+predictor_dropout: 0.5
+predictor_grad: 0.1
+predictor_hidden: -1
+predictor_kernel: 5
+predictor_layers: 2
+pretrain_fs_ckpt: ''
+print_nan_grads: false
+processed_data_dir: data/processed/LJSpeech
+profile_infer: false
+raw_data_dir: data/raw/LJSpeech
+ref_hidden_stride_kernel:
+- 0,3,5
+- 0,3,5
+- 0,2,5
+- 0,2,5
+- 0,2,5
+ref_level_db: 20
+ref_norm_layer: bn
+rename_tmux: true
+residual_channels: 256
+residual_layers: 20
+resume_from_checkpoint: 0
+save_best: true
+save_codes: []
+save_f0: false
+save_gt: true
+schedule_type: vpsde
+scheduler: rsqrt
+seed: 1234
+sil_add_noise: false
+sort_by_len: true
+spec_max: []
+spec_min: []
+task_cls: modules.ProDiff.task.ProDiff_teacher_task.ProDiff_teacher_Task
+tb_log_interval: 100
+test_ids: []
+test_input_dir: ''
+test_num: 100
+test_set_name: test
+timescale: 1
+timesteps: 4
+train_set_name: train
+train_sets: ''
+use_cond_disc: true
+use_energy_embed: true
+use_gt_dur: true
+use_gt_f0: true
+use_pitch_embed: true
+use_pos_embed: true
+use_ref_enc: false
+use_spk_embed: false
+use_spk_id: true
+use_split_spk_id: false
+use_uv: true
+use_var_enc: false
+val_check_interval: 2000
+valid_infer_interval: 10000
+valid_monitor_key: val_loss
+valid_monitor_mode: min
+valid_set_name: valid
+var_enc_vq_codes: 64
+vocoder_denoise_c: 0.0
+warmup_updates: 2000
+weight_decay: 0
+win_size: 1024
+word_size: 30000
+work_dir: checkpoints/ProDiff_Teacher1

checkpoints/ProDiff_Teacher/model_ckpt_steps_188000.ckpt

@@ -0,0 +1,3 @@
+version https://git-lfs.github.com/spec/v1
+oid sha256:5d3d02a215431c69dd54c1413b9a02cdc32795e2039ad9be857b12e85c470eea
+size 342252871

data_gen/tts/base_binarizer.py

@@ -0,0 +1,224 @@
+import os
+os.environ["OMP_NUM_THREADS"] = "1"
+
+from utils.multiprocess_utils import chunked_multiprocess_run
+import random
+import traceback
+import json
+from resemblyzer import VoiceEncoder
+from tqdm import tqdm
+from data_gen.tts.data_gen_utils import get_mel2ph, get_pitch, build_phone_encoder
+from utils.hparams import set_hparams, hparams
+import numpy as np
+from utils.indexed_datasets import IndexedDatasetBuilder
+from vocoders.base_vocoder import VOCODERS
+import pandas as pd
+
+
+class BinarizationError(Exception):
+    pass
+
+
+class BaseBinarizer:
+    def __init__(self, processed_data_dir=None):
+        if processed_data_dir is None:
+            processed_data_dir = hparams['processed_data_dir']
+        self.processed_data_dirs = processed_data_dir.split(",")
+        self.binarization_args = hparams['binarization_args']
+        self.pre_align_args = hparams['pre_align_args']
+        self.forced_align = self.pre_align_args['forced_align']
+        tg_dir = None
+        if self.forced_align == 'mfa':
+            tg_dir = 'mfa_outputs'
+        if self.forced_align == 'kaldi':
+            tg_dir = 'kaldi_outputs'
+        self.item2txt = {}
+        self.item2ph = {}
+        self.item2wavfn = {}
+        self.item2tgfn = {}
+        self.item2spk = {}
+        for ds_id, processed_data_dir in enumerate(self.processed_data_dirs):
+            self.meta_df = pd.read_csv(f"{processed_data_dir}/metadata_phone.csv", dtype=str)
+            for r_idx, r in self.meta_df.iterrows():
+                item_name = raw_item_name = r['item_name']
+                if len(self.processed_data_dirs) > 1:
+                    item_name = f'ds{ds_id}_{item_name}'
+                self.item2txt[item_name] = r['txt']
+                self.item2ph[item_name] = r['ph']
+                self.item2wavfn[item_name] = os.path.join(hparams['raw_data_dir'], 'wavs', os.path.basename(r['wav_fn']).split('_')[1])
+                self.item2spk[item_name] = r.get('spk', 'SPK1')
+                if len(self.processed_data_dirs) > 1:
+                    self.item2spk[item_name] = f"ds{ds_id}_{self.item2spk[item_name]}"
+                if tg_dir is not None:
+                    self.item2tgfn[item_name] = f"{processed_data_dir}/{tg_dir}/{raw_item_name}.TextGrid"
+        self.item_names = sorted(list(self.item2txt.keys()))
+        if self.binarization_args['shuffle']:
+            random.seed(1234)
+            random.shuffle(self.item_names)
+
+    @property
+    def train_item_names(self):
+        return self.item_names[hparams['test_num']+hparams['valid_num']:]
+
+    @property
+    def valid_item_names(self):
+        return self.item_names[0: hparams['test_num']+hparams['valid_num']]  #
+
+    @property
+    def test_item_names(self):
+        return self.item_names[0: hparams['test_num']]  # Audios for MOS testing are in 'test_ids'
+
+    def build_spk_map(self):
+        spk_map = set()
+        for item_name in self.item_names:
+            spk_name = self.item2spk[item_name]
+            spk_map.add(spk_name)
+        spk_map = {x: i for i, x in enumerate(sorted(list(spk_map)))}
+        assert len(spk_map) == 0 or len(spk_map) <= hparams['num_spk'], len(spk_map)
+        return spk_map
+
+    def item_name2spk_id(self, item_name):
+        return self.spk_map[self.item2spk[item_name]]
+
+    def _phone_encoder(self):
+        ph_set_fn = f"{hparams['binary_data_dir']}/phone_set.json"
+        ph_set = []
+        if hparams['reset_phone_dict'] or not os.path.exists(ph_set_fn):
+            for processed_data_dir in self.processed_data_dirs:
+                ph_set += [x.split(' ')[0] for x in open(f'{processed_data_dir}/dict.txt').readlines()]
+            ph_set = sorted(set(ph_set))
+            json.dump(ph_set, open(ph_set_fn, 'w'))
+        else:
+            ph_set = json.load(open(ph_set_fn, 'r'))
+        print("| phone set: ", ph_set)
+        return build_phone_encoder(hparams['binary_data_dir'])
+
+    def meta_data(self, prefix):
+        if prefix == 'valid':
+            item_names = self.valid_item_names
+        elif prefix == 'test':
+            item_names = self.test_item_names
+        else:
+            item_names = self.train_item_names
+        for item_name in item_names:
+            ph = self.item2ph[item_name]
+            txt = self.item2txt[item_name]
+            tg_fn = self.item2tgfn.get(item_name)
+            wav_fn = self.item2wavfn[item_name]
+            spk_id = self.item_name2spk_id(item_name)
+            yield item_name, ph, txt, tg_fn, wav_fn, spk_id
+
+    def process(self):
+        os.makedirs(hparams['binary_data_dir'], exist_ok=True)
+        self.spk_map = self.build_spk_map()
+        print("| spk_map: ", self.spk_map)
+        spk_map_fn = f"{hparams['binary_data_dir']}/spk_map.json"
+        json.dump(self.spk_map, open(spk_map_fn, 'w'))
+
+        self.phone_encoder = self._phone_encoder()
+        self.process_data('valid')
+        self.process_data('test')
+        self.process_data('train')
+
+    def process_data(self, prefix):
+        data_dir = hparams['binary_data_dir']
+        args = []
+        builder = IndexedDatasetBuilder(f'{data_dir}/{prefix}')
+        lengths = []
+        f0s = []
+        total_sec = 0
+        if self.binarization_args['with_spk_embed']:
+            voice_encoder = VoiceEncoder().cuda()
+
+        meta_data = list(self.meta_data(prefix))
+        for m in meta_data:
+            args.append(list(m) + [self.phone_encoder, self.binarization_args])
+        num_workers = int(os.getenv('N_PROC', os.cpu_count() // 3))
+        for f_id, (_, item) in enumerate(
+                zip(tqdm(meta_data), chunked_multiprocess_run(self.process_item, args, num_workers=num_workers))):
+            if item is None:
+                continue
+            item['spk_embed'] = voice_encoder.embed_utterance(item['wav']) \
+                if self.binarization_args['with_spk_embed'] else None
+            if not self.binarization_args['with_wav'] and 'wav' in item:
+                print("del wav")
+                del item['wav']
+            builder.add_item(item)
+            lengths.append(item['len'])
+            total_sec += item['sec']
+            if item.get('f0') is not None:
+                f0s.append(item['f0'])
+        builder.finalize()
+        np.save(f'{data_dir}/{prefix}_lengths.npy', lengths)
+        if len(f0s) > 0:
+            f0s = np.concatenate(f0s, 0)
+            f0s = f0s[f0s != 0]
+            np.save(f'{data_dir}/{prefix}_f0s_mean_std.npy', [np.mean(f0s).item(), np.std(f0s).item()])
+        print(f"| {prefix} total duration: {total_sec:.3f}s")
+
+    @classmethod
+    def process_item(cls, item_name, ph, txt, tg_fn, wav_fn, spk_id, encoder, binarization_args):
+        if hparams['vocoder'] in VOCODERS:
+            wav, mel = VOCODERS[hparams['vocoder']].wav2spec(wav_fn)
+        else:
+            wav, mel = VOCODERS[hparams['vocoder'].split('.')[-1]].wav2spec(wav_fn)
+        res = {
+            'item_name': item_name, 'txt': txt, 'ph': ph, 'mel': mel, 'wav': wav, 'wav_fn': wav_fn,
+            'sec': len(wav) / hparams['audio_sample_rate'], 'len': mel.shape[0], 'spk_id': spk_id
+        }
+        try:
+            if binarization_args['with_f0']:
+                cls.get_pitch(wav, mel, res)
+                if binarization_args['with_f0cwt']:
+                    cls.get_f0cwt(res['f0'], res)
+            if binarization_args['with_txt']:
+                try:
+                    phone_encoded = res['phone'] = encoder.encode(ph)
+                except:
+                    traceback.print_exc()
+                    raise BinarizationError(f"Empty phoneme")
+                if binarization_args['with_align']:
+                    cls.get_align(tg_fn, ph, mel, phone_encoded, res)
+        except BinarizationError as e:
+            print(f"| Skip item ({e}). item_name: {item_name}, wav_fn: {wav_fn}")
+            return None
+        return res
+
+    @staticmethod
+    def get_align(tg_fn, ph, mel, phone_encoded, res):
+        if tg_fn is not None and os.path.exists(tg_fn):
+            mel2ph, dur = get_mel2ph(tg_fn, ph, mel, hparams)
+        else:
+            raise BinarizationError(f"Align not found")
+        if mel2ph.max() - 1 >= len(phone_encoded):
+            raise BinarizationError(
+                f"Align does not match: mel2ph.max() - 1: {mel2ph.max() - 1}, len(phone_encoded): {len(phone_encoded)}")
+        res['mel2ph'] = mel2ph
+        res['dur'] = dur
+
+    @staticmethod
+    def get_pitch(wav, mel, res):
+        f0, pitch_coarse = get_pitch(wav, mel, hparams)
+        if sum(f0) == 0:
+            raise BinarizationError("Empty f0")
+        res['f0'] = f0
+        res['pitch'] = pitch_coarse
+
+    @staticmethod
+    def get_f0cwt(f0, res):
+        from utils.cwt import get_cont_lf0, get_lf0_cwt
+        uv, cont_lf0_lpf = get_cont_lf0(f0)
+        logf0s_mean_org, logf0s_std_org = np.mean(cont_lf0_lpf), np.std(cont_lf0_lpf)
+        cont_lf0_lpf_norm = (cont_lf0_lpf - logf0s_mean_org) / logf0s_std_org
+        Wavelet_lf0, scales = get_lf0_cwt(cont_lf0_lpf_norm)
+        if np.any(np.isnan(Wavelet_lf0)):
+            raise BinarizationError("NaN CWT")
+        res['cwt_spec'] = Wavelet_lf0
+        res['cwt_scales'] = scales
+        res['f0_mean'] = logf0s_mean_org
+        res['f0_std'] = logf0s_std_org
+
+
+if __name__ == "__main__":
+    set_hparams()
+    BaseBinarizer().process()

data_gen/tts/base_preprocess.py

@@ -0,0 +1,245 @@
+import json
+import os
+import random
+import re
+import traceback
+from collections import Counter
+from functools import partial
+
+import librosa
+from tqdm import tqdm
+from data_gen.tts.txt_processors.base_text_processor import get_txt_processor_cls
+from data_gen.tts.wav_processors.base_processor import get_wav_processor_cls
+from utils.hparams import hparams
+from utils.multiprocess_utils import multiprocess_run_tqdm
+from utils.os_utils import link_file, move_file, remove_file
+from data_gen.tts.data_gen_utils import is_sil_phoneme, build_token_encoder
+
+
+class BasePreprocessor:
+    def __init__(self):
+        self.preprocess_args = hparams['preprocess_args']
+        txt_processor = self.preprocess_args['txt_processor']
+        self.txt_processor = get_txt_processor_cls(txt_processor)
+        self.raw_data_dir = hparams['raw_data_dir']
+        self.processed_dir = hparams['processed_data_dir']
+        self.spk_map_fn = f"{self.processed_dir}/spk_map.json"
+
+    def meta_data(self):
+        """
+        :return: {'item_name': Str, 'wav_fn': Str, 'txt': Str, 'spk_name': Str, 'txt_loader': None or Func}
+        """
+        raise NotImplementedError
+
+    def process(self):
+        processed_dir = self.processed_dir
+        wav_processed_tmp_dir = f'{processed_dir}/processed_tmp'
+        remove_file(wav_processed_tmp_dir)
+        os.makedirs(wav_processed_tmp_dir, exist_ok=True)
+        wav_processed_dir = f'{processed_dir}/{self.wav_processed_dirname}'
+        remove_file(wav_processed_dir)
+        os.makedirs(wav_processed_dir, exist_ok=True)
+
+        meta_data = list(tqdm(self.meta_data(), desc='Load meta data'))
+        item_names = [d['item_name'] for d in meta_data]
+        assert len(item_names) == len(set(item_names)), 'Key `item_name` should be Unique.'
+
+        # preprocess data
+        phone_list = []
+        word_list = []
+        spk_names = set()
+        process_item = partial(self.preprocess_first_pass,
+                               txt_processor=self.txt_processor,
+                               wav_processed_dir=wav_processed_dir,
+                               wav_processed_tmp=wav_processed_tmp_dir,
+                               preprocess_args=self.preprocess_args)
+        items = []
+        args = [{
+            'item_name': item_raw['item_name'],
+            'txt_raw': item_raw['txt'],
+            'wav_fn': item_raw['wav_fn'],
+            'txt_loader': item_raw.get('txt_loader'),
+            'others': item_raw.get('others', None)
+        } for item_raw in meta_data]
+        for item_, (item_id, item) in zip(meta_data, multiprocess_run_tqdm(process_item, args, desc='Preprocess')):
+            if item is not None:
+                item_.update(item)
+                item = item_
+                if 'txt_loader' in item:
+                    del item['txt_loader']
+                item['id'] = item_id
+                item['spk_name'] = item.get('spk_name', '<SINGLE_SPK>')
+                item['others'] = item.get('others', None)
+                phone_list += item['ph'].split(" ")
+                word_list += item['word'].split(" ")
+                spk_names.add(item['spk_name'])
+                items.append(item)
+
+        # add encoded tokens
+        ph_encoder, word_encoder = self._phone_encoder(phone_list), self._word_encoder(word_list)
+        spk_map = self.build_spk_map(spk_names)
+        args = [{
+            'ph': item['ph'], 'word': item['word'], 'spk_name': item['spk_name'],
+            'word_encoder': word_encoder, 'ph_encoder': ph_encoder, 'spk_map': spk_map
+        } for item in items]
+        for idx, item_new_kv in multiprocess_run_tqdm(self.preprocess_second_pass, args, desc='Add encoded tokens'):
+            items[idx].update(item_new_kv)
+
+        # build mfa data
+        if self.preprocess_args['use_mfa']:
+            mfa_dict = set()
+            mfa_input_dir = f'{processed_dir}/mfa_inputs'
+            remove_file(mfa_input_dir)
+            # group MFA inputs for better parallelism
+            mfa_groups = [i // self.preprocess_args['nsample_per_mfa_group'] for i in range(len(items))]
+            if self.preprocess_args['mfa_group_shuffle']:
+                random.seed(hparams['seed'])
+                random.shuffle(mfa_groups)
+            args = [{
+                'item': item, 'mfa_input_dir': mfa_input_dir,
+                'mfa_group': mfa_group, 'wav_processed_tmp': wav_processed_tmp_dir,
+                'preprocess_args': self.preprocess_args
+            } for item, mfa_group in zip(items, mfa_groups)]
+            for i, (ph_gb_word_nosil, new_wav_align_fn) in multiprocess_run_tqdm(
+                    self.build_mfa_inputs, args, desc='Build MFA data'):
+                items[i]['wav_align_fn'] = new_wav_align_fn
+                for w in ph_gb_word_nosil.split(" "):
+                    mfa_dict.add(f"{w} {w.replace('_', ' ')}")
+            mfa_dict = sorted(mfa_dict)
+            with open(f'{processed_dir}/mfa_dict.txt', 'w') as f:
+                f.writelines([f'{l}\n' for l in mfa_dict])
+        with open(f"{processed_dir}/{self.meta_csv_filename}.json", 'w') as f:
+            f.write(re.sub(r'\n\s+([\d+\]])', r'\1', json.dumps(items, ensure_ascii=False, sort_keys=False, indent=1)))
+        remove_file(wav_processed_tmp_dir)
+
+    @classmethod
+    def preprocess_first_pass(cls, item_name, txt_raw, txt_processor,
+                              wav_fn, wav_processed_dir, wav_processed_tmp,
+                              preprocess_args, txt_loader=None, others=None):
+        try:
+            if txt_loader is not None:
+                txt_raw = txt_loader(txt_raw)
+            ph, txt, word, ph2word, ph_gb_word = cls.txt_to_ph(txt_processor, txt_raw, preprocess_args)
+            wav_fn, wav_align_fn = cls.process_wav(
+                item_name, wav_fn,
+                hparams['processed_data_dir'],
+                wav_processed_tmp, preprocess_args)
+
+            # wav for binarization
+            ext = os.path.splitext(wav_fn)[1]
+            os.makedirs(wav_processed_dir, exist_ok=True)
+            new_wav_fn = f"{wav_processed_dir}/{item_name}{ext}"
+            move_link_func = move_file if os.path.dirname(wav_fn) == wav_processed_tmp else link_file
+            move_link_func(wav_fn, new_wav_fn)
+            return {
+                'txt': txt, 'txt_raw': txt_raw, 'ph': ph,
+                'word': word, 'ph2word': ph2word, 'ph_gb_word': ph_gb_word,
+                'wav_fn': new_wav_fn, 'wav_align_fn': wav_align_fn,
+                'others': others
+            }
+        except:
+            traceback.print_exc()
+            print(f"| Error is caught. item_name: {item_name}.")
+            return None
+
+    @staticmethod
+    def txt_to_ph(txt_processor, txt_raw, preprocess_args):
+        txt_struct, txt = txt_processor.process(txt_raw, preprocess_args)
+        ph = [p for w in txt_struct for p in w[1]]
+        return " ".join(ph), txt
+
+    @staticmethod
+    def process_wav(item_name, wav_fn, processed_dir, wav_processed_tmp, preprocess_args):
+        processors = [get_wav_processor_cls(v) for v in preprocess_args['wav_processors']]
+        processors = [k() for k in processors if k is not None]
+        if len(processors) >= 1:
+            sr_file = librosa.core.get_samplerate(wav_fn)
+            output_fn_for_align = None
+            ext = os.path.splitext(wav_fn)[1]
+            input_fn = f"{wav_processed_tmp}/{item_name}{ext}"
+            link_file(wav_fn, input_fn)
+            for p in processors:
+                outputs = p.process(input_fn, sr_file, wav_processed_tmp, processed_dir, item_name, preprocess_args)
+                if len(outputs) == 3:
+                    input_fn, sr, output_fn_for_align = outputs
+                else:
+                    input_fn, sr = outputs
+            return input_fn, output_fn_for_align
+        else:
+            return wav_fn, wav_fn
+
+    def _phone_encoder(self, ph_set):
+        ph_set_fn = f"{self.processed_dir}/phone_set.json"
+        if self.preprocess_args['reset_phone_dict'] or not os.path.exists(ph_set_fn):
+            ph_set = sorted(set(ph_set))
+            json.dump(ph_set, open(ph_set_fn, 'w'), ensure_ascii=False)
+            print("| Build phone set: ", ph_set)
+        else:
+            ph_set = json.load(open(ph_set_fn, 'r'))
+            print("| Load phone set: ", ph_set)
+        return build_token_encoder(ph_set_fn)
+
+    def _word_encoder(self, word_set):
+        word_set_fn = f"{self.processed_dir}/word_set.json"
+        if self.preprocess_args['reset_word_dict']:
+            word_set = Counter(word_set)
+            total_words = sum(word_set.values())
+            word_set = word_set.most_common(hparams['word_dict_size'])
+            num_unk_words = total_words - sum([x[1] for x in word_set])
+            word_set = ['<BOS>', '<EOS>'] + [x[0] for x in word_set]
+            word_set = sorted(set(word_set))
+            json.dump(word_set, open(word_set_fn, 'w'), ensure_ascii=False)
+            print(f"| Build word set. Size: {len(word_set)}, #total words: {total_words},"
+                  f" #unk_words: {num_unk_words}, word_set[:10]:, {word_set[:10]}.")
+        else:
+            word_set = json.load(open(word_set_fn, 'r'))
+            print("| Load word set. Size: ", len(word_set), word_set[:10])
+        return build_token_encoder(word_set_fn)
+
+    @classmethod
+    def preprocess_second_pass(cls, word, ph, spk_name, word_encoder, ph_encoder, spk_map):
+        word_token = word_encoder.encode(word)
+        ph_token = ph_encoder.encode(ph)
+        spk_id = spk_map[spk_name]
+        return {'word_token': word_token, 'ph_token': ph_token, 'spk_id': spk_id}
+
+    def build_spk_map(self, spk_names):
+        spk_map = {x: i for i, x in enumerate(sorted(list(spk_names)))}
+        assert len(spk_map) == 0 or len(spk_map) <= hparams['num_spk'], len(spk_map)
+        print(f"| Number of spks: {len(spk_map)}, spk_map: {spk_map}")
+        json.dump(spk_map, open(self.spk_map_fn, 'w'), ensure_ascii=False)
+        return spk_map
+
+    @classmethod
+    def build_mfa_inputs(cls, item, mfa_input_dir, mfa_group, wav_processed_tmp, preprocess_args):
+        item_name = item['item_name']
+        wav_align_fn = item['wav_align_fn']
+        ph_gb_word = item['ph_gb_word']
+        ext = os.path.splitext(wav_align_fn)[1]
+        mfa_input_group_dir = f'{mfa_input_dir}/{mfa_group}'
+        os.makedirs(mfa_input_group_dir, exist_ok=True)
+        new_wav_align_fn = f"{mfa_input_group_dir}/{item_name}{ext}"
+        move_link_func = move_file if os.path.dirname(wav_align_fn) == wav_processed_tmp else link_file
+        move_link_func(wav_align_fn, new_wav_align_fn)
+        ph_gb_word_nosil = " ".join(["_".join([p for p in w.split("_") if not is_sil_phoneme(p)])
+                                     for w in ph_gb_word.split(" ") if not is_sil_phoneme(w)])
+        with open(f'{mfa_input_group_dir}/{item_name}.lab', 'w') as f_txt:
+            f_txt.write(ph_gb_word_nosil)
+        return ph_gb_word_nosil, new_wav_align_fn
+
+    def load_spk_map(self, base_dir):
+        spk_map_fn = f"{base_dir}/spk_map.json"
+        spk_map = json.load(open(spk_map_fn, 'r'))
+        return spk_map
+
+    def load_dict(self, base_dir):
+        ph_encoder = build_token_encoder(f'{base_dir}/phone_set.json')
+        return ph_encoder
+
+    @property
+    def meta_csv_filename(self):
+        return 'metadata'
+
+    @property
+    def wav_processed_dirname(self):
+        return 'wav_processed'

data_gen/tts/bin/binarize.py

@@ -0,0 +1,20 @@
+import os
+
+os.environ["OMP_NUM_THREADS"] = "1"
+
+import importlib
+from utils.hparams import set_hparams, hparams
+
+
+def binarize():
+    binarizer_cls = hparams.get("binarizer_cls", 'data_gen.tts.base_binarizer.BaseBinarizer')
+    pkg = ".".join(binarizer_cls.split(".")[:-1])
+    cls_name = binarizer_cls.split(".")[-1]
+    binarizer_cls = getattr(importlib.import_module(pkg), cls_name)
+    print("| Binarizer: ", binarizer_cls)
+    binarizer_cls().process()
+
+
+if __name__ == '__main__':
+    set_hparams()
+    binarize()

data_gen/tts/bin/pre_align.py

@@ -0,0 +1,20 @@
+import os
+
+os.environ["OMP_NUM_THREADS"] = "1"
+
+import importlib
+from utils.hparams import set_hparams, hparams
+
+
+def pre_align():
+    assert hparams['pre_align_cls'] != ''
+
+    pkg = ".".join(hparams["pre_align_cls"].split(".")[:-1])
+    cls_name = hparams["pre_align_cls"].split(".")[-1]
+    process_cls = getattr(importlib.import_module(pkg), cls_name)
+    process_cls().process()
+
+
+if __name__ == '__main__':
+    set_hparams()
+    pre_align()

data_gen/tts/bin/train_mfa_align.py

@@ -0,0 +1,15 @@
+import subprocess
+from utils.hparams import hparams, set_hparams
+import os
+
+
+def train_mfa_align():
+    CORPUS = hparams['processed_data_dir'].split("/")[-1]
+    print(f"| Run MFA for {CORPUS}.")
+    NUM_JOB = int(os.getenv('N_PROC', os.cpu_count()))
+    subprocess.check_call(f'CORPUS={CORPUS} NUM_JOB={NUM_JOB} bash usr/run_mfa_train_align.sh', shell=True)
+
+
+if __name__ == '__main__':
+    set_hparams(print_hparams=False)
+    train_mfa_align()

data_gen/tts/data_gen_utils.py

@@ -0,0 +1,352 @@
+import warnings
+
+warnings.filterwarnings("ignore")
+
+# import parselmouth
+import os
+import torch
+from skimage.transform import resize
+from utils.text_encoder import TokenTextEncoder
+from utils.pitch_utils import f0_to_coarse
+import struct
+import webrtcvad
+from scipy.ndimage.morphology import binary_dilation
+import librosa
+import numpy as np
+from utils import audio
+import pyloudnorm as pyln
+import re
+import json
+from collections import OrderedDict
+
+PUNCS = '!,.?;:'
+
+int16_max = (2 ** 15) - 1
+
+
+def trim_long_silences(path, sr=None, return_raw_wav=False, norm=True, vad_max_silence_length=12):
+    """
+    Ensures that segments without voice in the waveform remain no longer than a
+    threshold determined by the VAD parameters in params.py.
+    :param wav: the raw waveform as a numpy array of floats
+    :param vad_max_silence_length: Maximum number of consecutive silent frames a segment can have.
+    :return: the same waveform with silences trimmed away (length <= original wav length)
+    """
+
+    ## Voice Activation Detection
+    # Window size of the VAD. Must be either 10, 20 or 30 milliseconds.
+    # This sets the granularity of the VAD. Should not need to be changed.
+    sampling_rate = 16000
+    wav_raw, sr = librosa.core.load(path, sr=sr)
+
+    if norm:
+        meter = pyln.Meter(sr)  # create BS.1770 meter
+        loudness = meter.integrated_loudness(wav_raw)
+        wav_raw = pyln.normalize.loudness(wav_raw, loudness, -20.0)
+        if np.abs(wav_raw).max() > 1.0:
+            wav_raw = wav_raw / np.abs(wav_raw).max()
+
+    wav = librosa.resample(wav_raw, sr, sampling_rate, res_type='kaiser_best')
+
+    vad_window_length = 30  # In milliseconds
+    # Number of frames to average together when performing the moving average smoothing.
+    # The larger this value, the larger the VAD variations must be to not get smoothed out.
+    vad_moving_average_width = 8
+
+    # Compute the voice detection window size
+    samples_per_window = (vad_window_length * sampling_rate) // 1000
+
+    # Trim the end of the audio to have a multiple of the window size
+    wav = wav[:len(wav) - (len(wav) % samples_per_window)]
+
+    # Convert the float waveform to 16-bit mono PCM
+    pcm_wave = struct.pack("%dh" % len(wav), *(np.round(wav * int16_max)).astype(np.int16))
+
+    # Perform voice activation detection
+    voice_flags = []
+    vad = webrtcvad.Vad(mode=3)
+    for window_start in range(0, len(wav), samples_per_window):
+        window_end = window_start + samples_per_window
+        voice_flags.append(vad.is_speech(pcm_wave[window_start * 2:window_end * 2],
+                                         sample_rate=sampling_rate))
+    voice_flags = np.array(voice_flags)
+
+    # Smooth the voice detection with a moving average
+    def moving_average(array, width):
+        array_padded = np.concatenate((np.zeros((width - 1) // 2), array, np.zeros(width // 2)))
+        ret = np.cumsum(array_padded, dtype=float)
+        ret[width:] = ret[width:] - ret[:-width]
+        return ret[width - 1:] / width
+
+    audio_mask = moving_average(voice_flags, vad_moving_average_width)
+    audio_mask = np.round(audio_mask).astype(np.bool)
+
+    # Dilate the voiced regions
+    audio_mask = binary_dilation(audio_mask, np.ones(vad_max_silence_length + 1))
+    audio_mask = np.repeat(audio_mask, samples_per_window)
+    audio_mask = resize(audio_mask, (len(wav_raw),)) > 0
+    if return_raw_wav:
+        return wav_raw, audio_mask, sr
+    return wav_raw[audio_mask], audio_mask, sr
+
+
+def process_utterance(wav_path,
+                      fft_size=1024,
+                      hop_size=256,
+                      win_length=1024,
+                      window="hann",
+                      num_mels=80,
+                      fmin=80,
+                      fmax=7600,
+                      eps=1e-6,
+                      sample_rate=22050,
+                      loud_norm=False,
+                      min_level_db=-100,
+                      return_linear=False,
+                      trim_long_sil=False, vocoder='pwg'):
+    if isinstance(wav_path, str):
+        if trim_long_sil:
+            wav, _, _ = trim_long_silences(wav_path, sample_rate)
+        else:
+            wav, _ = librosa.core.load(wav_path, sr=sample_rate)
+    else:
+        wav = wav_path
+
+    if loud_norm:
+        meter = pyln.Meter(sample_rate)  # create BS.1770 meter
+        loudness = meter.integrated_loudness(wav)
+        wav = pyln.normalize.loudness(wav, loudness, -22.0)
+        if np.abs(wav).max() > 1:
+            wav = wav / np.abs(wav).max()
+
+    # get amplitude spectrogram
+    x_stft = librosa.stft(wav, n_fft=fft_size, hop_length=hop_size,
+                          win_length=win_length, window=window, pad_mode="constant")
+    spc = np.abs(x_stft)  # (n_bins, T)
+
+    # get mel basis
+    fmin = 0 if fmin == -1 else fmin
+    fmax = sample_rate / 2 if fmax == -1 else fmax
+    mel_basis = librosa.filters.mel(sample_rate, fft_size, num_mels, fmin, fmax)
+    mel = mel_basis @ spc
+
+    if vocoder == 'pwg':
+        mel = np.log10(np.maximum(eps, mel))  # (n_mel_bins, T)
+    else:
+        assert False, f'"{vocoder}" is not in ["pwg"].'
+
+    l_pad, r_pad = audio.librosa_pad_lr(wav, fft_size, hop_size, 1)
+    wav = np.pad(wav, (l_pad, r_pad), mode='constant', constant_values=0.0)
+    wav = wav[:mel.shape[1] * hop_size]
+
+    if not return_linear:
+        return wav, mel
+    else:
+        spc = audio.amp_to_db(spc)
+        spc = audio.normalize(spc, {'min_level_db': min_level_db})
+        return wav, mel, spc
+
+
+def get_pitch(wav_data, mel, hparams):
+    """
+
+    :param wav_data: [T]
+    :param mel: [T, 80]
+    :param hparams:
+    :return:
+    """
+    time_step = hparams['hop_size'] / hparams['audio_sample_rate'] * 1000
+    f0_min = 80
+    f0_max = 750
+
+    if hparams['hop_size'] == 128:
+        pad_size = 4
+    elif hparams['hop_size'] == 256:
+        pad_size = 2
+    else:
+        assert False
+
+    f0 = parselmouth.Sound(wav_data, hparams['audio_sample_rate']).to_pitch_ac(
+        time_step=time_step / 1000, voicing_threshold=0.6,
+        pitch_floor=f0_min, pitch_ceiling=f0_max).selected_array['frequency']
+    lpad = pad_size * 2
+    rpad = len(mel) - len(f0) - lpad
+    f0 = np.pad(f0, [[lpad, rpad]], mode='constant')
+    # mel and f0 are extracted by 2 different libraries. we should force them to have the same length.
+    # Attention: we find that new version of some libraries could cause ``rpad'' to be a negetive value...
+    # Just to be sure, we recommend users to set up the same environments as them in requirements_auto.txt (by Anaconda)
+    delta_l = len(mel) - len(f0)
+    assert np.abs(delta_l) <= 8
+    if delta_l > 0:
+        f0 = np.concatenate([f0, [f0[-1]] * delta_l], 0)
+    f0 = f0[:len(mel)]
+    pitch_coarse = f0_to_coarse(f0)
+    return f0, pitch_coarse
+
+
+def remove_empty_lines(text):
+    """remove empty lines"""
+    assert (len(text) > 0)
+    assert (isinstance(text, list))
+    text = [t.strip() for t in text]
+    if "" in text:
+        text.remove("")
+    return text
+
+
+class TextGrid(object):
+    def __init__(self, text):
+        text = remove_empty_lines(text)
+        self.text = text
+        self.line_count = 0
+        self._get_type()
+        self._get_time_intval()
+        self._get_size()
+        self.tier_list = []
+        self._get_item_list()
+
+    def _extract_pattern(self, pattern, inc):
+        """
+        Parameters
+        ----------
+        pattern : regex to extract pattern
+        inc : increment of line count after extraction
+        Returns
+        -------
+        group : extracted info
+        """
+        try:
+            group = re.match(pattern, self.text[self.line_count]).group(1)
+            self.line_count += inc
+        except AttributeError:
+            raise ValueError("File format error at line %d:%s" % (self.line_count, self.text[self.line_count]))
+        return group
+
+    def _get_type(self):
+        self.file_type = self._extract_pattern(r"File type = \"(.*)\"", 2)
+
+    def _get_time_intval(self):
+        self.xmin = self._extract_pattern(r"xmin = (.*)", 1)
+        self.xmax = self._extract_pattern(r"xmax = (.*)", 2)
+
+    def _get_size(self):
+        self.size = int(self._extract_pattern(r"size = (.*)", 2))
+
+    def _get_item_list(self):
+        """Only supports IntervalTier currently"""
+        for itemIdx in range(1, self.size + 1):
+            tier = OrderedDict()
+            item_list = []
+            tier_idx = self._extract_pattern(r"item \[(.*)\]:", 1)
+            tier_class = self._extract_pattern(r"class = \"(.*)\"", 1)
+            if tier_class != "IntervalTier":
+                raise NotImplementedError("Only IntervalTier class is supported currently")
+            tier_name = self._extract_pattern(r"name = \"(.*)\"", 1)
+            tier_xmin = self._extract_pattern(r"xmin = (.*)", 1)
+            tier_xmax = self._extract_pattern(r"xmax = (.*)", 1)
+            tier_size = self._extract_pattern(r"intervals: size = (.*)", 1)
+            for i in range(int(tier_size)):
+                item = OrderedDict()
+                item["idx"] = self._extract_pattern(r"intervals \[(.*)\]", 1)
+                item["xmin"] = self._extract_pattern(r"xmin = (.*)", 1)
+                item["xmax"] = self._extract_pattern(r"xmax = (.*)", 1)
+                item["text"] = self._extract_pattern(r"text = \"(.*)\"", 1)
+                item_list.append(item)
+            tier["idx"] = tier_idx
+            tier["class"] = tier_class
+            tier["name"] = tier_name
+            tier["xmin"] = tier_xmin
+            tier["xmax"] = tier_xmax
+            tier["size"] = tier_size
+            tier["items"] = item_list
+            self.tier_list.append(tier)
+
+    def toJson(self):
+        _json = OrderedDict()
+        _json["file_type"] = self.file_type
+        _json["xmin"] = self.xmin
+        _json["xmax"] = self.xmax
+        _json["size"] = self.size
+        _json["tiers"] = self.tier_list
+        return json.dumps(_json, ensure_ascii=False, indent=2)
+
+
+def get_mel2ph(tg_fn, ph, mel, hparams):
+    ph_list = ph.split(" ")
+    with open(tg_fn, "r") as f:
+        tg = f.readlines()
+    tg = remove_empty_lines(tg)
+    tg = TextGrid(tg)
+    tg = json.loads(tg.toJson())
+    split = np.ones(len(ph_list) + 1, np.float) * -1
+    tg_idx = 0
+    ph_idx = 0
+    tg_align = [x for x in tg['tiers'][-1]['items']]
+    tg_align_ = []
+    for x in tg_align:
+        x['xmin'] = float(x['xmin'])
+        x['xmax'] = float(x['xmax'])
+        if x['text'] in ['sil', 'sp', '', 'SIL', 'PUNC']:
+            x['text'] = ''
+            if len(tg_align_) > 0 and tg_align_[-1]['text'] == '':
+                tg_align_[-1]['xmax'] = x['xmax']
+                continue
+        tg_align_.append(x)
+    tg_align = tg_align_
+    tg_len = len([x for x in tg_align if x['text'] != ''])
+    ph_len = len([x for x in ph_list if not is_sil_phoneme(x)])
+    assert tg_len == ph_len, (tg_len, ph_len, tg_align, ph_list, tg_fn)
+    while tg_idx < len(tg_align) or ph_idx < len(ph_list):
+        if tg_idx == len(tg_align) and is_sil_phoneme(ph_list[ph_idx]):
+            split[ph_idx] = 1e8
+            ph_idx += 1
+            continue
+        x = tg_align[tg_idx]
+        if x['text'] == '' and ph_idx == len(ph_list):
+            tg_idx += 1
+            continue
+        assert ph_idx < len(ph_list), (tg_len, ph_len, tg_align, ph_list, tg_fn)
+        ph = ph_list[ph_idx]
+        if x['text'] == '' and not is_sil_phoneme(ph):
+            assert False, (ph_list, tg_align)
+        if x['text'] != '' and is_sil_phoneme(ph):
+            ph_idx += 1
+        else:
+            assert (x['text'] == '' and is_sil_phoneme(ph)) \
+                   or x['text'].lower() == ph.lower() \
+                   or x['text'].lower() == 'sil', (x['text'], ph)
+            split[ph_idx] = x['xmin']
+            if ph_idx > 0 and split[ph_idx - 1] == -1 and is_sil_phoneme(ph_list[ph_idx - 1]):
+                split[ph_idx - 1] = split[ph_idx]
+            ph_idx += 1
+            tg_idx += 1
+    assert tg_idx == len(tg_align), (tg_idx, [x['text'] for x in tg_align])
+    assert ph_idx >= len(ph_list) - 1, (ph_idx, ph_list, len(ph_list), [x['text'] for x in tg_align], tg_fn)
+    mel2ph = np.zeros([mel.shape[0]], np.int)
+    split[0] = 0
+    split[-1] = 1e8
+    for i in range(len(split) - 1):
+        assert split[i] != -1 and split[i] <= split[i + 1], (split[:-1],)
+    split = [int(s * hparams['audio_sample_rate'] / hparams['hop_size'] + 0.5) for s in split]
+    for ph_idx in range(len(ph_list)):
+        mel2ph[split[ph_idx]:split[ph_idx + 1]] = ph_idx + 1
+    mel2ph_torch = torch.from_numpy(mel2ph)
+    T_t = len(ph_list)
+    dur = mel2ph_torch.new_zeros([T_t + 1]).scatter_add(0, mel2ph_torch, torch.ones_like(mel2ph_torch))
+    dur = dur[1:].numpy()
+    return mel2ph, dur
+
+
+def build_phone_encoder(data_dir):
+    phone_list_file = os.path.join(data_dir, 'phone_set.json')
+    phone_list = json.load(open(phone_list_file))
+    return TokenTextEncoder(None, vocab_list=phone_list, replace_oov=',')
+
+
+def is_sil_phoneme(p):
+    return not p[0].isalpha()
+
+
+def build_token_encoder(token_list_file):
+    token_list = json.load(open(token_list_file))
+    return TokenTextEncoder(None, vocab_list=token_list, replace_oov='<UNK>')

data_gen/tts/txt_processors/__init__.py

@@ -0,0 +1 @@
+from . import en

data_gen/tts/txt_processors/base_text_processor.py

@@ -0,0 +1,47 @@
+from data_gen.tts.data_gen_utils import is_sil_phoneme
+
+REGISTERED_TEXT_PROCESSORS = {}
+
+def register_txt_processors(name):
+    def _f(cls):
+        REGISTERED_TEXT_PROCESSORS[name] = cls
+        return cls
+
+    return _f
+
+
+def get_txt_processor_cls(name):
+    return REGISTERED_TEXT_PROCESSORS.get(name, None)
+
+
+class BaseTxtProcessor:
+    @staticmethod
+    def sp_phonemes():
+        return ['|']
+
+    @classmethod
+    def process(cls, txt, preprocess_args):
+        raise NotImplementedError
+
+    @classmethod
+    def postprocess(cls, txt_struct, preprocess_args):
+        # remove sil phoneme in head and tail
+        while len(txt_struct) > 0 and is_sil_phoneme(txt_struct[0][0]):
+            txt_struct = txt_struct[1:]
+        while len(txt_struct) > 0 and is_sil_phoneme(txt_struct[-1][0]):
+            txt_struct = txt_struct[:-1]
+        if preprocess_args['with_phsep']:
+            txt_struct = cls.add_bdr(txt_struct)
+        if preprocess_args['add_eos_bos']:
+            txt_struct = [["<BOS>", ["<BOS>"]]] + txt_struct + [["<EOS>", ["<EOS>"]]]
+        return txt_struct
+
+    @classmethod
+    def add_bdr(cls, txt_struct):
+        txt_struct_ = []
+        for i, ts in enumerate(txt_struct):
+            txt_struct_.append(ts)
+            if i != len(txt_struct) - 1 and \
+                    not is_sil_phoneme(txt_struct[i][0]) and not is_sil_phoneme(txt_struct[i + 1][0]):
+                txt_struct_.append(['|', ['|']])
+        return txt_struct_

data_gen/tts/txt_processors/en.py

@@ -0,0 +1,77 @@
+import re
+import unicodedata
+
+from g2p_en import G2p
+from g2p_en.expand import normalize_numbers
+from nltk import pos_tag
+from nltk.tokenize import TweetTokenizer
+
+from data_gen.tts.txt_processors.base_text_processor import BaseTxtProcessor, register_txt_processors
+from data_gen.tts.data_gen_utils import is_sil_phoneme, PUNCS
+
+class EnG2p(G2p):
+    word_tokenize = TweetTokenizer().tokenize
+
+    def __call__(self, text):
+        # preprocessing
+        words = EnG2p.word_tokenize(text)
+        tokens = pos_tag(words)  # tuples of (word, tag)
+
+        # steps
+        prons = []
+        for word, pos in tokens:
+            if re.search("[a-z]", word) is None:
+                pron = [word]
+
+            elif word in self.homograph2features:  # Check homograph
+                pron1, pron2, pos1 = self.homograph2features[word]
+                if pos.startswith(pos1):
+                    pron = pron1
+                else:
+                    pron = pron2
+            elif word in self.cmu:  # lookup CMU dict
+                pron = self.cmu[word][0]
+            else:  # predict for oov
+                pron = self.predict(word)
+
+            prons.extend(pron)
+            prons.extend([" "])
+
+        return prons[:-1]
+
+
+@register_txt_processors('en')
+class TxtProcessor(BaseTxtProcessor):
+    g2p = EnG2p()
+
+    @staticmethod
+    def preprocess_text(text):
+        text = normalize_numbers(text)
+        text = ''.join(char for char in unicodedata.normalize('NFD', text)
+                       if unicodedata.category(char) != 'Mn')  # Strip accents
+        text = text.lower()
+        text = re.sub("[\'\"()]+", "", text)
+        text = re.sub("[-]+", " ", text)
+        text = re.sub(f"[^ a-z{PUNCS}]", "", text)
+        text = re.sub(f" ?([{PUNCS}]) ?", r"\1", text)  # !! -> !
+        text = re.sub(f"([{PUNCS}])+", r"\1", text)  # !! -> !
+        text = text.replace("i.e.", "that is")
+        text = text.replace("i.e.", "that is")
+        text = text.replace("etc.", "etc")
+        text = re.sub(f"([{PUNCS}])", r" \1 ", text)
+        text = re.sub(rf"\s+", r" ", text)
+        return text
+
+    @classmethod
+    def process(cls, txt, preprocess_args):
+        txt = cls.preprocess_text(txt).strip()
+        phs = cls.g2p(txt)
+        txt_struct = [[w, []] for w in txt.split(" ")]
+        i_word = 0
+        for p in phs:
+            if p == ' ':
+                i_word += 1
+            else:
+                txt_struct[i_word][1].append(p)
+        txt_struct = cls.postprocess(txt_struct, preprocess_args)
+        return txt_struct, txt

data_gen/tts/wav_processors/__init__.py

@@ -0,0 +1,2 @@
+from . import base_processor
+from . import common_processors

data_gen/tts/wav_processors/base_processor.py

@@ -0,0 +1,25 @@
+REGISTERED_WAV_PROCESSORS = {}
+
+
+def register_wav_processors(name):
+    def _f(cls):
+        REGISTERED_WAV_PROCESSORS[name] = cls
+        return cls
+
+    return _f
+
+
+def get_wav_processor_cls(name):
+    return REGISTERED_WAV_PROCESSORS.get(name, None)
+
+
+class BaseWavProcessor:
+    @property
+    def name(self):
+        raise NotImplementedError
+
+    def output_fn(self, input_fn):
+        return f'{input_fn[:-4]}_{self.name}.wav'
+
+    def process(self, input_fn, sr, tmp_dir, processed_dir, item_name, preprocess_args):
+        raise NotImplementedError

data_gen/tts/wav_processors/common_processors.py

@@ -0,0 +1,86 @@
+import os
+import subprocess
+import librosa
+import numpy as np
+from data_gen.tts.wav_processors.base_processor import BaseWavProcessor, register_wav_processors
+from data_gen.tts.data_gen_utils import trim_long_silences
+from utils.audio import save_wav
+from utils.rnnoise import rnnoise
+from utils.hparams import hparams
+
+
+@register_wav_processors(name='sox_to_wav')
+class ConvertToWavProcessor(BaseWavProcessor):
+    @property
+    def name(self):
+        return 'ToWav'
+
+    def process(self, input_fn, sr, tmp_dir, processed_dir, item_name, preprocess_args):
+        if input_fn[-4:] == '.wav':
+            return input_fn, sr
+        else:
+            output_fn = self.output_fn(input_fn)
+            subprocess.check_call(f'sox -v 0.95 "{input_fn}" -t wav "{output_fn}"', shell=True)
+            return output_fn, sr
+
+
+@register_wav_processors(name='sox_resample')
+class ResampleProcessor(BaseWavProcessor):
+    @property
+    def name(self):
+        return 'Resample'
+
+    def process(self, input_fn, sr, tmp_dir, processed_dir, item_name, preprocess_args):
+        output_fn = self.output_fn(input_fn)
+        sr_file = librosa.core.get_samplerate(input_fn)
+        if sr != sr_file:
+            subprocess.check_call(f'sox -v 0.95 "{input_fn}" -r{sr} "{output_fn}"', shell=True)
+            y, _ = librosa.core.load(input_fn, sr=sr)
+            y, _ = librosa.effects.trim(y)
+            save_wav(y, output_fn, sr)
+            return output_fn, sr
+        else:
+            return input_fn, sr
+
+
+@register_wav_processors(name='trim_sil')
+class TrimSILProcessor(BaseWavProcessor):
+    @property
+    def name(self):
+        return 'TrimSIL'
+
+    def process(self, input_fn, sr, tmp_dir, processed_dir, item_name, preprocess_args):
+        output_fn = self.output_fn(input_fn)
+        y, _ = librosa.core.load(input_fn, sr=sr)
+        y, _ = librosa.effects.trim(y)
+        save_wav(y, output_fn, sr)
+        return output_fn
+
+
+@register_wav_processors(name='trim_all_sil')
+class TrimAllSILProcessor(BaseWavProcessor):
+    @property
+    def name(self):
+        return 'TrimSIL'
+
+    def process(self, input_fn, sr, tmp_dir, processed_dir, item_name, preprocess_args):
+        output_fn = self.output_fn(input_fn)
+        y, audio_mask, _ = trim_long_silences(
+            input_fn, vad_max_silence_length=preprocess_args.get('vad_max_silence_length', 12))
+        save_wav(y, output_fn, sr)
+        if preprocess_args['save_sil_mask']:
+            os.makedirs(f'{processed_dir}/sil_mask', exist_ok=True)
+            np.save(f'{processed_dir}/sil_mask/{item_name}.npy', audio_mask)
+        return output_fn, sr
+
+
+@register_wav_processors(name='denoise')
+class DenoiseProcessor(BaseWavProcessor):
+    @property
+    def name(self):
+        return 'Denoise'
+
+    def process(self, input_fn, sr, tmp_dir, processed_dir, item_name, preprocess_args):
+        output_fn = self.output_fn(input_fn)
+        rnnoise(input_fn, output_fn, out_sample_rate=sr)
+        return output_fn, sr